similar to: R: Asterisk and Call Hold

Hi folks, in order to establish backward compatibility we made an image that automatically detects if the other side does not support RFC3264. Please try it out, we would be very interested if this image is a progress! http://snom.com/download/share/snom200-2.03b-SIP.bin Thanks, CS

Hi all! I updated my snom200 to 2.02t and now MOH from * don?t works anymore... only the MOH from snom server and if i clear the MOH server field in the phone i have no MOH at all..( with the transfer button, moh plays using a extension). Someone with that problem? I downgrade to 2.01s but nothing changes. Miklos -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I think I met a case similar to the one solved by [1] . Quoting this case : * res_pjsip: Handle deferred SDP hold/unhold properly. Some SIP devices indicate hold/unhold using deferred SDP reinvites. In other words, they provide no SDP in the reinvite. A typical transaction that starts hold might look something like this: * Device sends reinvite with no SDP * Asterisk

Hi all If Asterisk is bridging a call between two SIP peers and one peer puts the other on hold by means of a re-INVITE with SDP containing a=sendonly, Asterisk will play locally generated MOH instead of relaying the media streamed by the SIP peer which took the hold action. Any ideas how to change that? (This is understandable if the peer is a handset but can be a problem if it is a PBX with

Hi All, When asterisk bridges a call between 2 peers and peer-A's user puts the call on hold, then peer-A sends a INVITE with recvonly in the SDP. Asterisk responds to peer-A with sendonly in the SDP and asterisk sends an INVITE to peer-B with recvonly in the SDP. Peer-B then responds with a sendonly in the SDP. I've noticed in the above scenario that peer-B contiutes to send audio to

I'm working through my list of features people will expect, and Hold Pickup is at the top at the moment -- has anyone done any work on this? We've had some unpleasant experiences with call parking, and everyone seems to like the Hold Pickup model. If you don't know what I mean by Hold Pickup, it's sort of a reverse transfer; pick up the nearest phone and dial

On 12.12.18 at 19:43 Joshua C. Colp wrote: > On Wed, Dec 12, 2018, at 12:31 PM, Michael Maier wrote: > > <snip> > >> >> The problem: The extension doesn't create a ringback locally, because >> it most probably expects it to >> be sent by the callee - but the callee doesn't send anything (not >> surprising, because there has been >>

Hello all, I have a problem where problem with one way audio, and I think it's related to "a=sendonly" and a re-invite. Can anyone please assist? The scenario is as follows.... - We send an INVITE to a peer, and it replies with a "100 Trying", and then a "183 Session Progress" message containing "a=sendonly". - Asterisk plays the caller music on hold,

Hi masters, I?m not an expert on this my friends, but I?m trying to understand which the expected behaviour is from Asterisk side when you deal with the following scenario: Caller ?> GSM Gateway with SIM card A ?> Asterisk queue ?> extension 1000 GSM gateway with call waiting activated on SIM A Queue with ?skip busy agent? disabled and ringall strategy. SIP extension 1000 with call

Hi, Recently we got a new feature request from our customer, they want a report to list the duration that agents putting customer on hold, they want to base on this to measure the agents performance. I cannot find any events in cdr, message logs, or manager interface, only when I enable sip debug, then I can see the ReInvite Event in the cli , some thing like the attached logs, is there any

Hello, I'm running Dovecot on a FreeBSD system with Postfix in a virtual user setup, with Mysql. I am trying to understand the quota configuration. I've got a Mysql database with an accounts table with a quota field. I've also got two other tables one quota (currently has nothing in it an empty set), and quota2 messages and bytes which has one entry. My goal is to have different

Hello everybody, I'm having issues with calls being dropped on Aastra phones, when the call is on hold. Tested with models 6863i and 6867i. I've figured that the call is dropped by Asterisk when it reaches the rtpholdtimeout limit. I've reported the issue to Aastra, asking them to implement some kind of "RTP keep-alive" feature on their phones. Maybe the phone could send

Hi! Maybe I am missing something or am a little blind at the moment, but I didn't find out how asterisk can place a call on hold when acting as user agent client to another SIP server. Scenario: ---------- Asterisk registers to another SIP server (provider) as user agent. An inbound call from this other SIP server comes in and arrives at asterisk. Asterisk performs some actions in the

Hi Is it a normal behavior of Asterisk put a call on hold when receive a Session Progress with media address 0.0.0.0 in SDP? I believe the call on hold should be initiate with a re-invite. Thanks -- Att, Rafael Saraiva -------------- next part -------------- An HTML attachment was scrubbed... URL:

hi, its possibe to "dont start" music on hold when caller (from sip operator trunk) press HOLD (i.e. on mobile phone) Asterisk acts on SDP a=sendonly i want pass trough media from SIP trunk provider Marek

Hello, I am having problems with music on hold on grandstream phones. When I press Hold button on grandstream phone this is the debug of sip. But nothing happens, no music. Is it problem of asterisk or grandstream budget phone? Sip read: INVITE sip:1105@192.168.1.1 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK7fcd3a44e7721b41 From:

Good evening, I am Marta Colombo, student at Milan's Politecnico. Thank you very much for your kindness, this mailing list is really useful. I am using the function kde2d for two-dimensional kernel density estimation and I'd like to know something more about this kind of density estimator. In particular I'd like to know: what bandwidth is used ? Thank you in advance for your attention

Good evening,I am Marta Colombo, student of "Politecnico di Milano". I'm studying Local Regression Techniques such as loess, smoothing splines and kernel smoothers. Choosing "symmetric" for the argument "family" in loess function it is possible to produce a robust estimate , in function smooth.spline and ksmooth I didn't find this possibility. Well, is there a

Hello, I've been racking my brain over this for much of the day so I thought the list would probably be more helpful. A few days ago I upgraded from Asterisk 1.2 to Asterisk 1.4.5. Everything appeared to be working properly. However, on the first business day, we realized that when transferring calls (not using call parking, using the built in transfer buttons on a Cisco 7960) would not

Hi R, A quick question: How do we find the number of words in a string? Example: C="Have a nice day" And the number of words should be 4. any built in function or?... Thanks, Shubha Shubha Karanth | Amba Research Ph +91 80 3980 8031 | Mob +91 94 4886 4510 Bangalore * Colombo * London * New York * San José * Singapore * www.ambaresearch.com This e-mail may contain